Merge branch 'fix/pcm-jiffies-check' into topic/pcm-jiffies-check

31 files changed, 225 insertions, 103 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 7fbd68fab944..5c48e36038f2 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -1074,7 +1074,7 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
         return i;
 }
 
-static int __devinit aaci_probe(struct amba_device *dev, void *id)
+static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
 {
         struct aaci *aaci;
         int ret, i;
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index a2c12d105c9a..6fdca97186e7 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action)
                 switch (resetgpio_action) {
                 case RESETGPIO_NORMAL_ALTFUNC:
                         if (reset_gpio == 113)
-                                mode = 113 | GPIO_OUT | GPIO_DFLT_LOW;
+                                mode = 113 | GPIO_ALT_FN_2_OUT;
                         if (reset_gpio == 95)
                                 mode = 95 | GPIO_ALT_FN_1_OUT;
                         break;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 63d088f2265f..d659995ac3ac 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -249,6 +249,17 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
                         new_hw_ptr = hw_base + pos;
                 }
         }
+
+        /* Do jiffies check only in xrun_debug mode */
+        if (!xrun_debug(substream))
+                goto no_jiffies_check;
+
+        /* Skip the jiffies check for hardwares with BATCH flag.
+         * Such hardware usually just increases the position at each IRQ,
+         * thus it can't give any strange position.
+         */
+        if (runtime->hw.info & SNDRV_PCM_INFO_BATCH)
+                goto no_jiffies_check;
         hdelta = new_hw_ptr - old_hw_ptr;
         jdelta = jiffies - runtime->hw_ptr_jiffies;
         if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) {
@@ -272,6 +283,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
                 hw_base -= hw_base % runtime->buffer_size;
                 delta = 0;
         }
+ no_jiffies_check:
         if (delta > runtime->period_size + runtime->period_size / 2) {
                 hw_ptr_error(substream,
                              "Lost interrupts? "
@@ -329,7 +341,9 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
                         hw_base = 0;
                 new_hw_ptr = hw_base + pos;
         }
-        if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) {
+        /* Do jiffies check only in xrun_debug mode */
+        if (xrun_debug(substream) &&
+            ((delta * HZ) / runtime->rate) > jdelta + HZ/100) {
                 hw_ptr_error(substream,
                              "hw_ptr skipping! "
                              "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n",
@@ -1471,7 +1485,6 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream,
                 runtime->status->hw_ptr %= runtime->buffer_size;
         else
                 runtime->status->hw_ptr = 0;
-        runtime->hw_ptr_jiffies = jiffies;
         snd_pcm_stream_unlock_irqrestore(substream, flags);
         return 0;
 }
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index cb769d415db7..45dc53fcfa2f 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -849,6 +849,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state)
 {
         struct snd_pcm_runtime *runtime = substream->runtime;
         snd_pcm_trigger_tstamp(substream);
+        runtime->hw_ptr_jiffies = jiffies;
         runtime->status->state = state;
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
             runtime->silence_size > 0)
@@ -962,6 +963,11 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push)
 {
         if (substream->runtime->trigger_master != substream)
                 return 0;
+        /* The jiffies check in snd_pcm_update_hw_ptr*() is done by
+         * a delta betwen the current jiffies, this gives a large enough
+         * delta, effectively to skip the check once.
+         */
+        substream->runtime->hw_ptr_jiffies = jiffies - HZ * 1000;
         return substream->ops->trigger(substream,
                                        push ? SNDRV_PCM_TRIGGER_PAUSE_PUSH :
                                               SNDRV_PCM_TRIGGER_PAUSE_RELEASE);
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index caeb0f57fcca..199b03377142 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -50,8 +50,8 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
         uinfo->value.enumerated.items = chip->max_treble + 1;
         if (uinfo->value.enumerated.item > chip->max_treble)
                 uinfo->value.enumerated.item = chip->max_treble;
-        sprintf(uinfo->value.enumerated.name, "%d",
-                        PCSP_CALC_RATE(uinfo->value.enumerated.item));
+        sprintf(uinfo->value.enumerated.name, "%lu",
+                (unsigned long)PCSP_CALC_RATE(uinfo->value.enumerated.item));
         return 0;
 }
 
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index b2b6d50c9425..a25fb7b1f441 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -963,16 +963,11 @@ static int __devinit snd_serial_probe(struct platform_device *devptr)
         if (err < 0)
                 goto _err;
 
-        sprintf(card->longname, "%s at 0x%lx, irq %d speed %d div %d outs %d ins %d adaptor %s droponfull %d",
+        sprintf(card->longname, "%s [%s] at %#lx, irq %d",
                 card->shortname,
-                uart->base,
-                uart->irq,
-                uart->speed,
-                (int)uart->divisor,
-                outs[dev],
-                ins[dev],
                 adaptor_names[uart->adaptor],
-                uart->drop_on_full);
+                uart->base,
+                uart->irq);
 
         snd_card_set_dev(card, &devptr->dev);
 
diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c
index 906454413ed2..3a1526ae1729 100644
--- a/sound/isa/msnd/msnd.c
+++ b/sound/isa/msnd/msnd.c
@@ -438,7 +438,8 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip,
 static struct snd_pcm_hardware snd_msnd_playback = {
         .info =                 SNDRV_PCM_INFO_MMAP |
                                 SNDRV_PCM_INFO_INTERLEAVED |
-                                SNDRV_PCM_INFO_MMAP_VALID,
+                                SNDRV_PCM_INFO_MMAP_VALID |
+                                SNDRV_PCM_INFO_BATCH,
         .formats =              SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
         .rates =                SNDRV_PCM_RATE_8000_48000,
         .rate_min =             8000,
@@ -456,7 +457,8 @@ static struct snd_pcm_hardware snd_msnd_playback = {
 static struct snd_pcm_hardware snd_msnd_capture = {
         .info =                 SNDRV_PCM_INFO_MMAP |
                                 SNDRV_PCM_INFO_INTERLEAVED |
-                                SNDRV_PCM_INFO_MMAP_VALID,
+                                SNDRV_PCM_INFO_MMAP_VALID |
+                                SNDRV_PCM_INFO_BATCH,
         .formats =              SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
         .rates =                SNDRV_PCM_RATE_8000_48000,
         .rate_min =             8000,
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 81bc93e5f1e3..7337abdbe4e3 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -958,10 +958,13 @@ static int patch_sigmatel_stac9708_3d(struct snd_ac97 * ac97)
 }
 
 static const struct snd_kcontrol_new snd_ac97_sigmatel_4speaker =
-AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
+AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch",
+                AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
 
+/* "Sigmatel " removed due to excessive name length: */
 static const struct snd_kcontrol_new snd_ac97_sigmatel_phaseinvert =
-AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
+AC97_SINGLE("Surround Phase Inversion Playback Switch",
+                AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
 
 static const struct snd_kcontrol_new snd_ac97_sigmatel_controls[] = {
 AC97_SINGLE("Sigmatel DAC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 1, 1, 0),
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index a299340519df..ce3f2e90f4d7 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -349,7 +349,8 @@ static struct snd_pcm_hardware snd_bt87x_digital_hw = {
         .info = SNDRV_PCM_INFO_MMAP |
                 SNDRV_PCM_INFO_INTERLEAVED |
                 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-                SNDRV_PCM_INFO_MMAP_VALID,
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_BATCH,
         .formats = SNDRV_PCM_FMTBIT_S16_LE,
         .rates = 0, /* set at runtime */
         .channels_min = 2,
@@ -365,7 +366,8 @@ static struct snd_pcm_hardware snd_bt87x_analog_hw = {
         .info = SNDRV_PCM_INFO_MMAP |
                 SNDRV_PCM_INFO_INTERLEAVED |
                 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-                SNDRV_PCM_INFO_MMAP_VALID,
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_BATCH,
         .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
         .rates = SNDRV_PCM_RATE_KNOT,
         .rate_min = ANALOG_CLOCK / CLOCK_DIV_MAX,
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index ad2888705d2a..c111efe61c3c 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -800,7 +800,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
                 "Capture Volume",
                 "External Amplifier",
                 "Sigmatel 4-Speaker Stereo Playback Switch",
-                "Sigmatel Surround Phase Inversion Playback ",
+                "Surround Phase Inversion Playback Switch",
                 NULL
         };
         static char *ca0106_rename_ctls[] = {
diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c
index 3482ef69f491..2e44316530a2 100644
--- a/sound/pci/echoaudio/indigodjx.c
+++ b/sound/pci/echoaudio/indigodjx.c
@@ -88,6 +88,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = {
         .rates =        SNDRV_PCM_RATE_32000 |
                         SNDRV_PCM_RATE_44100 |
                         SNDRV_PCM_RATE_48000 |
+                        SNDRV_PCM_RATE_64000 |
                         SNDRV_PCM_RATE_88200 |
                         SNDRV_PCM_RATE_96000,
         .rate_min = 32000,
diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c
index aebee27a40ff..eb3819f9654a 100644
--- a/sound/pci/echoaudio/indigoiox.c
+++ b/sound/pci/echoaudio/indigoiox.c
@@ -89,6 +89,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = {
         .rates =        SNDRV_PCM_RATE_32000 |
                         SNDRV_PCM_RATE_44100 |
                         SNDRV_PCM_RATE_48000 |
+                        SNDRV_PCM_RATE_64000 |
                         SNDRV_PCM_RATE_88200 |
                         SNDRV_PCM_RATE_96000,
         .rate_min = 32000,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b8a0d3e79272..bcbb736f94f0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12058,6 +12058,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
         SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
         SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
+        SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA),
         SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
         SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
         SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 917bc5d3ac2c..03b3646018a1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4079,7 +4079,12 @@ static int stac92xx_init(struct hda_codec *codec)
                                 pinctl = snd_hda_codec_read(codec, nid, 0,
                                         AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
                                 /* if PINCTL already set then skip */
-                                if (!(pinctl & AC_PINCTL_IN_EN)) {
+                                /* Also, if both INPUT and OUTPUT are set,
+                                 * it must be a BIOS bug; need to override, too
+                                 */
+                                if (!(pinctl & AC_PINCTL_IN_EN) ||
+                                    (pinctl & AC_PINCTL_OUT_EN)) {
+                                        pinctl &= ~AC_PINCTL_OUT_EN;
                                         pinctl |= AC_PINCTL_IN_EN;
                                         stac92xx_auto_set_pinctl(codec, nid,
                                                                  pinctl);
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 8b79969034be..7cc38a11e997 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -1238,7 +1238,8 @@ static struct snd_pcm_hardware snd_korg1212_playback_info =
 {
         .info =              (SNDRV_PCM_INFO_MMAP |
                               SNDRV_PCM_INFO_MMAP_VALID |
-                              SNDRV_PCM_INFO_INTERLEAVED),
+                              SNDRV_PCM_INFO_INTERLEAVED |
+                              SNDRV_PCM_INFO_BATCH),
         .formats =            SNDRV_PCM_FMTBIT_S16_LE,
         .rates =              (SNDRV_PCM_RATE_44100 |
                               SNDRV_PCM_RATE_48000),
@@ -1258,7 +1259,8 @@ static struct snd_pcm_hardware snd_korg1212_capture_info =
 {
         .info =              (SNDRV_PCM_INFO_MMAP |
                               SNDRV_PCM_INFO_MMAP_VALID |
-                              SNDRV_PCM_INFO_INTERLEAVED),
+                              SNDRV_PCM_INFO_INTERLEAVED |
+                              SNDRV_PCM_INFO_BATCH),
         .formats =            SNDRV_PCM_FMTBIT_S16_LE,
         .rates =              (SNDRV_PCM_RATE_44100 |
                               SNDRV_PCM_RATE_48000),
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 6f1034417a02..e51a5ef1954d 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -889,7 +889,7 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm,
         spin_lock_irqsave(&cif->lock, irqflags);
         while (i++ < CMDIF_TIMEOUT && !IS_READY(cif->hwport))
                 udelay(10);
-        if (i >= CMDIF_TIMEOUT) {
+        if (i > CMDIF_TIMEOUT) {
                 err = -EBUSY;
                 goto errout;
         }
@@ -907,8 +907,10 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm,
                         WRITE_PORT_ULONG(cmdport->data1, cmd);  /* write cmd */
                         if ((flags & RESP) && ret) {
                                 while (!IS_DATF(cmdport) &&
-                                       time++ < CMDIF_TIMEOUT)
+                                       time < CMDIF_TIMEOUT) {
                                         udelay(10);
+                                        time++;
+                                }
                                 if (time < CMDIF_TIMEOUT) {     /* read response */
                                         ret->retlongs[0] =
                                             READ_PORT_ULONG(cmdport->data1);
@@ -1454,7 +1456,7 @@ static int snd_riptide_trigger(struct snd_pcm_substream *substream, int cmd)
                         SEND_GPOS(cif, 0, data->id, &rptr);
                         udelay(1);
                 } while (i != rptr.retlongs[1] && j++ < MAX_WRITE_RETRY);
-                if (j >= MAX_WRITE_RETRY)
+                if (j > MAX_WRITE_RETRY)
                         snd_printk(KERN_ERR "Riptide: Could not stop stream!");
                 break;
         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
@@ -1783,7 +1785,7 @@ snd_riptide_codec_write(struct snd_ac97 *ac97, unsigned short reg,
                 SEND_SACR(cif, val, reg);
                 SEND_RACR(cif, reg, &rptr);
         } while (rptr.retwords[1] != val && i++ < MAX_WRITE_RETRY);
-        if (i == MAX_WRITE_RETRY)
+        if (i > MAX_WRITE_RETRY)
                 snd_printdd("Write AC97 reg failed\n");
 }
 
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 809b233dd4a3..1ef58c51c213 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1687,7 +1687,7 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol,
         return change;
 }
 
-static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -9450, 150, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -4650, 150, 1);
 
 static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = {
         .name = "PCM Playback Volume",
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index 01066c95580e..d057e6489643 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -240,7 +240,8 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs)
 static struct snd_pcm_hardware pdacf_pcm_capture_hw = {
         .info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
                                  SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
-                                 SNDRV_PCM_INFO_MMAP_VALID),
+                                 SNDRV_PCM_INFO_MMAP_VALID |
+                                 SNDRV_PCM_INFO_BATCH),
         .formats =              SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
                                 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
                                 SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a259148..594c6c5b7838 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
 /* PCM hardware DMA capabilities - platform specific */
 static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
         .info             = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-                            SNDRV_PCM_INFO_INTERLEAVED,
+                            SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
         .formats          = AU1XPSC_PCM_FMTS,
         .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
         .period_bytes_max = 4096 * 1024 - 1,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205de28a..df7c8c281d2f 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
 static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
 
 /*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
  * Capture gain after the ADCs
  * from 0 dB to 31 dB in 1 dB steps
  */
@@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
                 4, 3, 0, output_tvl),
 
         SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
-                TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+                TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
 
         /* Common capture gain controls */
         SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d0993bed9..0275321ff8ab 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
          * required for LRC in master mode. The DACs or ADCs need a
          * valid audio path i.e. pin -> ADC or DAC -> pin before
          * the LRC will be enabled in master mode. */
-        if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+        if (!master || cmd != SNDRV_PCM_TRIGGER_START)
                 return 0;
 
         if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c518c3e5aa3f..40cd274eb1ef 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
         inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 /* AINLMUX */
-SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
         &wm8990_dapm_ainlmux_controls, inmixer_event,
         SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
@@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
         inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 /* AINRMUX */
-SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
         &wm8990_dapm_ainrmux_controls, inmixer_event,
         SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
@@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = {
         {"LIN12 PGA", "LIN2 Switch", "LIN2"},
         /* LIN34 PGA */
         {"LIN34 PGA", "LIN3 Switch", "LIN3"},
-        {"LIN34 PGA", "LIN4 Switch", "LIN4"},
+        {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
         /* INMIXL */
         {"INMIXL", "Record Left Volume", "LOMIX"},
         {"INMIXL", "LIN2 Volume", "LIN2"},
         {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
         {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
-        /* AILNMUX */
-        {"AILNMUX", "INMIXL Mix", "INMIXL"},
-        {"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
-        {"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
-        {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
-        {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+        /* AINLMUX */
+        {"AINLMUX", "INMIXL Mix", "INMIXL"},
+        {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"},
+        {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"},
+        {"AINLMUX", "RXVOICE Mix", "LIN4/RXN"},
+        {"AINLMUX", "RXVOICE Mix", "RIN4/RXP"},
         /* ADC */
-        {"Left ADC", NULL, "AILNMUX"},
+        {"Left ADC", NULL, "AINLMUX"},
 
         /* RIN12 PGA */
         {"RIN12 PGA", "RIN1 Switch", "RIN1"},
         {"RIN12 PGA", "RIN2 Switch", "RIN2"},
         /* RIN34 PGA */
         {"RIN34 PGA", "RIN3 Switch", "RIN3"},
-        {"RIN34 PGA", "RIN4 Switch", "RIN4"},
+        {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
         /* INMIXL */
         {"INMIXR", "Record Right Volume", "ROMIX"},
         {"INMIXR", "RIN2 Volume", "RIN2"},
         {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
         {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
-        /* AIRNMUX */
-        {"AIRNMUX", "INMIXR Mix", "INMIXR"},
-        {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
-        {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
-        {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
-        {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+        /* AINRMUX */
+        {"AINRMUX", "INMIXR Mix", "INMIXR"},
+        {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"},
+        {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"},
+        {"AINRMUX", "RXVOICE Mix", "LIN4/RXN"},
+        {"AINRMUX", "RXVOICE Mix", "RIN4/RXP"},
         /* ADC */
-        {"Right ADC", NULL, "AIRNMUX"},
+        {"Right ADC", NULL, "AINRMUX"},
 
         /* LOMIX */
         {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
@@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
         {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
 
         /* OUT3MIX */
-        {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+        {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
         {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
 
         /* OUT4MIX */
@@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
         /* Output Pins */
         {"LON", NULL, "LONMIX"},
         {"LOP", NULL, "LOPMIX"},
-        {"OUT", NULL, "OUT3MIX"},
+        {"OUT3", NULL, "OUT3MIX"},
         {"LOUT", NULL, "LOUT PGA"},
         {"SPKN", NULL, "SPKMIX"},
         {"ROUT", NULL, "ROUT PGA"},
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index bd7392c9657e..411a710be660 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S
         tristate
 
 config SND_DAVINCI_SOC_EVM
-        tristate "SoC Audio support for DaVinci EVM"
-        depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM
+        tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+        depends on SND_DAVINCI_SOC
+        depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
         select SND_DAVINCI_SOC_I2S
         select SND_SOC_TLV320AIC3X
         help
           Say Y if you want to add support for SoC audio on TI
-          DaVinci EVM platform.
+          DaVinci DM6446 or DM355 EVM platforms.
 
 config SND_DAVINCI_SOC_SFFSDR
         tristate "SoC Audio support for SFFSDR"
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9b90b347007c..58fd1cbedd88 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -20,7 +20,11 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/dma.h>
-#include <mach/hardware.h>
+#include <asm/mach-types.h>
+
+#include <mach/asp.h>
+#include <mach/edma.h>
+#include <mach/mux.h>
 
 #include "../codecs/tlv320aic3x.h"
 #include "davinci-pcm.h"
@@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = {
 
 /* evm audio private data */
 static struct aic3x_setup_data evm_aic3x_setup = {
-        .i2c_bus = 0,
+        .i2c_bus = 1,
         .i2c_address = 0x1b,
 };
 
@@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = {
         .codec_data = &evm_aic3x_setup,
 };
 
+/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */
 static struct resource evm_snd_resources[] = {
         {
-                .start = DAVINCI_MCBSP_BASE,
-                .end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+                .start = DAVINCI_ASP0_BASE,
+                .end = DAVINCI_ASP0_BASE + SZ_8K - 1,
                 .flags = IORESOURCE_MEM,
         },
 };
 
 static struct evm_snd_platform_data evm_snd_data = {
-        .tx_dma_ch      = DM644X_DMACH_MCBSP_TX,
-        .rx_dma_ch      = DM644X_DMACH_MCBSP_RX,
+        .tx_dma_ch      = DAVINCI_DMA_ASP0_TX,
+        .rx_dma_ch      = DAVINCI_DMA_ASP0_RX,
+};
+
+/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */
+static struct resource dm335evm_snd_resources[] = {
+        {
+                .start = DAVINCI_ASP1_BASE,
+                .end = DAVINCI_ASP1_BASE + SZ_8K - 1,
+                .flags = IORESOURCE_MEM,
+        },
+};
+
+static struct evm_snd_platform_data dm335evm_snd_data = {
+        .tx_dma_ch      = DAVINCI_DMA_ASP1_TX,
+        .rx_dma_ch      = DAVINCI_DMA_ASP1_RX,
 };
 
 static struct platform_device *evm_snd_device;
 
 static int __init evm_init(void)
 {
+        struct resource *resources;
+        unsigned num_resources;
+        struct evm_snd_platform_data *data;
+        int index;
         int ret;
 
-        evm_snd_device = platform_device_alloc("soc-audio", 0);
+        if (machine_is_davinci_evm()) {
+                davinci_cfg_reg(DM644X_MCBSP);
+
+                resources = evm_snd_resources;
+                num_resources = ARRAY_SIZE(evm_snd_resources);
+                data = &evm_snd_data;
+                index = 0;
+        } else if (machine_is_davinci_dm355_evm()) {
+                /* we don't use ASP1 IRQs, or we'd need to mux them ... */
+                davinci_cfg_reg(DM355_EVT8_ASP1_TX);
+                davinci_cfg_reg(DM355_EVT9_ASP1_RX);
+
+                resources = dm335evm_snd_resources;
+                num_resources = ARRAY_SIZE(dm335evm_snd_resources);
+                data = &dm335evm_snd_data;
+                index = 1;
+        } else
+                return -EINVAL;
+
+        evm_snd_device = platform_device_alloc("soc-audio", index);
         if (!evm_snd_device)
                 return -ENOMEM;
 
         platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
         evm_snd_devdata.dev = &evm_snd_device->dev;
-        platform_device_add_data(evm_snd_device, &evm_snd_data,
-                                 sizeof(evm_snd_data));
+        platform_device_add_data(evm_snd_device, data, sizeof(*data));
 
-        ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
-                                            ARRAY_SIZE(evm_snd_resources));
+        ret = platform_device_add_resources(evm_snd_device, resources,
+                        num_resources);
         if (ret) {
                 platform_device_put(evm_snd_device);
                 return ret;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ffdb9439d3d8..b1ea52fc83c7 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -24,6 +24,26 @@
 
 #include "davinci-pcm.h"
 
+
+/*
+ * NOTE:  terminology here is confusing.
+ *
+ *  - This driver supports the "Audio Serial Port" (ASP),
+ *    found on dm6446, dm355, and other DaVinci chips.
+ *
+ *  - But it labels it a "Multi-channel Buffered Serial Port"
+ *    (McBSP) as on older chips like the dm642 ... which was
+ *    backward-compatible, possibly explaining that confusion.
+ *
+ *  - OMAP chips have a controller called McBSP, which is
+ *    incompatible with the DaVinci flavor of McBSP.
+ *
+ *  - Newer DaVinci chips have a controller called McASP,
+ *    incompatible with ASP and with either McBSP.
+ *
+ * In short:  this uses ASP to implement I2S, not McBSP.
+ * And it won't be the only DaVinci implemention of I2S.
+ */
 #define DAVINCI_MCBSP_DRR_REG   0x00
 #define DAVINCI_MCBSP_DXR_REG   0x04
 #define DAVINCI_MCBSP_SPCR_REG  0x08
@@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_card *card = socdev->card;
-        struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+        struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
         struct davinci_mcbsp_dev *dev;
         struct resource *mem, *ioarea;
         struct evm_snd_platform_data *pdata;
@@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 
         cpu_dai->private_data = dev;
 
-        dev->clk = clk_get(&pdev->dev, "McBSPCLK");
+        dev->clk = clk_get(&pdev->dev, NULL);
         if (IS_ERR(dev->clk)) {
                 ret = -ENODEV;
                 goto err_free_mem;
@@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_card *card = socdev->card;
-        struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+        struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
         struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
         struct resource *mem;
 
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 7af3b5b3a53d..a05996588489 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -22,6 +22,7 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
+#include <mach/edma.h>
 
 #include "davinci-pcm.h"
 
@@ -51,7 +52,7 @@ struct davinci_runtime_data {
         spinlock_t lock;
         int period;             /* current DMA period */
         int master_lch;         /* Master DMA channel */
-        int slave_lch;          /* Slave DMA channel */
+        int slave_lch;          /* linked parameter RAM reload slot */
         struct davinci_pcm_dma_params *params;  /* DMA params */
 };
 
@@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
                 dst_bidx = data_type;
         }
 
-        davinci_set_dma_src_params(lch, src, INCR, W8BIT);
-        davinci_set_dma_dest_params(lch, dst, INCR, W8BIT);
-        davinci_set_dma_src_index(lch, src_bidx, 0);
-        davinci_set_dma_dest_index(lch, dst_bidx, 0);
-        davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+        edma_set_src(lch, src, INCR, W8BIT);
+        edma_set_dest(lch, dst, INCR, W8BIT);
+        edma_set_src_index(lch, src_bidx, 0);
+        edma_set_dest_index(lch, dst_bidx, 0);
+        edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC);
 
         prtd->period++;
         if (unlikely(prtd->period >= runtime->periods))
                 prtd->period = 0;
 }
 
-static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
+static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
 {
         struct snd_pcm_substream *substream = data;
         struct davinci_runtime_data *prtd = substream->runtime->private_data;
@@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
         struct davinci_runtime_data *prtd = substream->runtime->private_data;
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
-        int tcc = TCC_ANY;
+        struct edmacc_param p_ram;
         int ret;
 
         if (!dma_data)
@@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
         prtd->params = dma_data;
 
         /* Request master DMA channel */
-        ret = davinci_request_dma(prtd->params->channel, prtd->params->name,
+        ret = edma_alloc_channel(prtd->params->channel,
                                   davinci_pcm_dma_irq, substream,
-                                  &prtd->master_lch, &tcc, EVENTQ_0);
-        if (ret)
+                                  EVENTQ_0);
+        if (ret < 0)
                 return ret;
+        prtd->master_lch = ret;
 
-        /* Request slave DMA channel */
-        ret = davinci_request_dma(PARAM_ANY, "Link",
-                                  NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0);
-        if (ret) {
-                davinci_free_dma(prtd->master_lch);
+        /* Request parameter RAM reload slot */
+        ret = edma_alloc_slot(EDMA_SLOT_ANY);
+        if (ret < 0) {
+                edma_free_channel(prtd->master_lch);
                 return ret;
         }
-
-        /* Link slave DMA channel in loopback */
-        davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch);
+        prtd->slave_lch = ret;
+
+        /* Issue transfer completion IRQ when the channel completes a
+         * transfer, then always reload from the same slot (by a kind
+         * of loopback link).  The completion IRQ handler will update
+         * the reload slot with a new buffer.
+         *
+         * REVISIT save p_ram here after setting up everything except
+         * the buffer and its length (ccnt) ... use it as a template
+         * so davinci_pcm_enqueue_dma() takes less time in IRQ.
+         */
+        edma_read_slot(prtd->slave_lch, &p_ram);
+        p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch);
+        p_ram.link_bcntrld = prtd->slave_lch << 5;
+        edma_write_slot(prtd->slave_lch, &p_ram);
 
         return 0;
 }
@@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
         case SNDRV_PCM_TRIGGER_START:
         case SNDRV_PCM_TRIGGER_RESUME:
         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-                davinci_start_dma(prtd->master_lch);
+                edma_start(prtd->master_lch);
                 break;
         case SNDRV_PCM_TRIGGER_STOP:
         case SNDRV_PCM_TRIGGER_SUSPEND:
         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-                davinci_stop_dma(prtd->master_lch);
+                edma_stop(prtd->master_lch);
                 break;
         default:
                 ret = -EINVAL;
@@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
 {
         struct davinci_runtime_data *prtd = substream->runtime->private_data;
-        struct paramentry_descriptor temp;
+        struct edmacc_param temp;
 
         prtd->period = 0;
         davinci_pcm_enqueue_dma(substream);
 
-        /* Get slave channel dma params for master channel startup */
-        davinci_get_dma_params(prtd->slave_lch, &temp);
-        davinci_set_dma_params(prtd->master_lch, &temp);
+        /* Copy self-linked parameter RAM entry into master channel */
+        edma_read_slot(prtd->slave_lch, &temp);
+        edma_write_slot(prtd->master_lch, &temp);
 
         return 0;
 }
@@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream)
 
         spin_lock(&prtd->lock);
 
-        davinci_dma_getposition(prtd->master_lch, &src, &dst);
+        edma_get_position(prtd->master_lch, &src, &dst);
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
                 count = src - runtime->dma_addr;
         else
@@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream)
         struct snd_pcm_runtime *runtime = substream->runtime;
         struct davinci_runtime_data *prtd = runtime->private_data;
 
-        davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch);
+        edma_unlink(prtd->slave_lch);
 
-        davinci_free_dma(prtd->slave_lch);
-        davinci_free_dma(prtd->master_lch);
+        edma_free_slot(prtd->slave_lch);
+        edma_free_channel(prtd->master_lch);
 
         kfree(prtd);
 
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729df27b5..1111c710118a 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 
 static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
         .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-                SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+                SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_BATCH,
         .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
                    SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
         .rate_min = 8000,
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 0dad3a0bb920..baddb1242c71 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = {
         .info = (SNDRV_PCM_INFO_MMAP |
                 SNDRV_PCM_INFO_INTERLEAVED |
                 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-                SNDRV_PCM_INFO_MMAP_VALID),
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_BATCH),
         .formats =      DMABRG_FMTS,
         .rates =        DMABRG_RATES,
         .rate_min =             8000,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 99712f652d0d..1cd149b9ce69 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev)
         struct snd_soc_platform *platform = card->platform;
         struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
+        if (!card->instantiated)
+                return 0;
+
         run_delayed_work(&card->delayed_work);
 
         if (platform->remove)
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index af95ff1e126c..1d2e51b3f918 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -1975,7 +1975,8 @@ static struct snd_pcm_hardware snd_dbri_pcm_hw = {
         .info           = SNDRV_PCM_INFO_MMAP |
                           SNDRV_PCM_INFO_INTERLEAVED |
                           SNDRV_PCM_INFO_BLOCK_TRANSFER |
-                          SNDRV_PCM_INFO_MMAP_VALID,
+                          SNDRV_PCM_INFO_MMAP_VALID |
+                          SNDRV_PCM_INFO_BATCH,
         .formats        = SNDRV_PCM_FMTBIT_MU_LAW |
                           SNDRV_PCM_FMTBIT_A_LAW |
                           SNDRV_PCM_FMTBIT_U8 |
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 9a608fa85155..dd1ab6177840 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -870,7 +870,8 @@ static struct snd_pcm_hardware snd_usX2Y_2c =
 {
         .info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
                                  SNDRV_PCM_INFO_BLOCK_TRANSFER |
-                                 SNDRV_PCM_INFO_MMAP_VALID),
+                                 SNDRV_PCM_INFO_MMAP_VALID |
+                                 SNDRV_PCM_INFO_BATCH),
         .formats =                 SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE,
         .rates =                   SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
         .rate_min =                44100,